The present invention generally relates to mode handling in the field of communication systems and, more particularly, to determining coding modes in digital communication systems that support multiple speech/forward error correction coding schemes.
The growth of commercial communication systems and, in particular, the explosive growth of cellular radiotelephone systems, have compelled system designers to search for ways to increase system capacity without reducing communication quality beyond consumer tolerance thresholds. One technique to achieve these objectives involved changing from systems wherein analog modulation was used to impress data onto a carrier wave, to systems wherein digital modulation was used to impress the data on carrier waves.
In wireless digital communication systems, standardized air interfaces specify most of the system parameters, including speech coding type(s), burst format, communication protocol, etc. For example, the European Telecommunication Standard Institute (ETSI) has specified a Global System for Mobile Communications (GSM) standard that uses time division multiple access (TDMA) to communicate control, voice and data information over radio frequency (RF) physical channels or links using a Gaussian Minimum Shift Keying (GMSK) modulation scheme at a symbol rate of 271 ksps. In the U.S., the Telecommunication Industry Association (TIA) has published a number of Interim Standards, such as IS-54 and IS-136, that define various versions of digital advanced mobile phone service (D-AMPS), a TDMA system that uses a differential quadrature phase shift keying (DQPSK) modulation scheme for communicating data over RF links.
TDMA systems subdivide the available frequency into one or more RF channels. The RF channels are further divided into a number of physical channels corresponding to timeslots in TDMA frames. Logical channels are formed of one or several physical channels where modulation and coding is specified. In these systems, the mobile stations communicate with a plurality of scattered base stations by transmitting and receiving bursts of digital information over uplink and downlink RF channels.
The growing number of mobile stations in use today has generated the need for more voice and data channels within cellular telecommunication systems. As a result, base stations have become more closely spaced, with an increase in interference between mobile stations operating on the same frequency in neighboring or closely spaced cells. In fact, some systems now employ code division multiple access (CDMA), using a form of spread spectrum modulation wherein signals intentionally share the same time and frequency. Although digital techniques provide a greater number of useful channels from a given frequency spectrum, there still remains a need to maintain interference at acceptable levels, or more specifically to monitor and control the ratio of the carrier signal strength to interference, (i.e., carrier-to-interference (C/I) ratio).
Another factor which is increasingly important in providing various communication services is the desired/required user bit rate for data to be transmitted over a particular connection. For example, for voice and/or data services, user bit rate corresponds to voice quality and/or data throughput, with a higher user bit rate producing better voice quality and/or higher data throughput. The total user bit rate is determined by a selected combination of techniques for speech coding, channel coding, modulation, and resource allocation, e.g., for a TDMA system, this latter technique may refer to the number of assignable time slots per connection, for a CDMA system, this latter parameter may refer to the number of assignable codes per connection.
Speech coding (or more generally "source coding") techniques are used to compress the input information into a format which uses an acceptable amount of bandwidth but from which an intelligible output signal can be reproduced. Many different types of speech coding algorithms exist, e.g., residual excited linear predictive (RELP), regular-pulse excitation (RPE), etc., the details of which are not particularly relevant to this invention. More significant in this context is the fact that various speech coders have various output bit rates and that, as one would expect, speech coders having a higher output bit rate tend to provide greater consumer acceptance of their reproduced voice quality than those having a lower output bit rate. As an example, consider that more traditional, wire-based telephone systems use PCM speech coding at 64 kbps, while GSM systems employ an RPE speech coding scheme operating at 13 kbps.
In addition to speech coding, digital communication systems also employ various techniques to handle erroneously received information. Generally speaking, these techniques include those which aid a receiver to correct the erroneously received information, e.g., forward error correction (FEC) techniques, and those which enable the erroneously received information to be retransmitted to the receiver, e.g., automatic retransmission request (ARQ) techniques. FEC techniques include, for example, convolutional or block coding (collectively referred to herein as "channel coding") of the data prior to modulation. Channel coding involves representing a certain number of data bits using a certain number of code bits. Thus, for example, it is common to refer to convolutional codes by their code rates, e.g., 1/2 and 1/3, wherein the lower code rates provide greater error protection but lower user bit rates for a given channel bit rate.
Conventionally, each of the techniques which impacted the user bit rate were fixed for any given radiocommunication system, or at least for the duration of a connection established by a radiocommunication system. That is, each system established connections that operated with one type of speech coding, one type of channel coding, one type of modulation and one resource allocation. More recently, however, dynamic adaptation of these techniques has become a popular method for optimizing system performance in the face of the numerous parameters which may vary rapidly over time, e.g., the radio propagation characteristics of radiocommunication channels, the loading of the system, the user's bit rate requirements, etc.
For example, different modulations have been dynamically assigned to selectively take advantage of the strengths of individual modulation schemes and to provide greater user bit rates and/or increased resistance to noise and interference. An example of a communication system employing multiple modulation schemes is found in U.S. Pat. No. 5,577,087. Therein, a technique for switching between 16 QAM and QPSK is described. The decision to switch between modulation types is made based on quality measurements, however this system employs a constant user bit rate which means that a change in modulation scheme also requires a change in channel bit rate, e.g., the number of timeslots used to support a transmission channel.
It is envisioned that many different combinations of these processing techniques may be selectively employed both as between different connections supported by a radiocommunication system and during the lifetime of a single connection. However, the receiver must be aware of the types of processing being used by the transmitter in order to properly decode the information upon receipt. Generally, there are two categories of techniques for informing a receiver about processing techniques associated with a radio signal: (1) explicit information, i.e., a message field within the transmitted information having a mode value that is indicative of the processing type(s) and (2) implicit information, which is sometimes referred to as "blind" decoding, whereupon the receiver determines the processing performed by the transmitter by analyzing the received signal. This latter technique is employed in CDMA systems operating in accordance with the TIA/EIA IS-95 standard. Explicit information is sometimes considered to be preferable because it reduces processing delay at the receiver, but comes at the cost of the need for the transmitter to include additional overhead bits along with the user data.
Of particular interest for the present invention are mode indicators which reflect the transmitter's currently employed speech coding/channel coding combination. For example, when channel conditions are good, the transmitter may employ a speech coding/channel coding mode which provides for a high source coding bit rate and a relatively low degree of error protection. Alternatively, when channel conditions are poor, then a coding mode which provides a low bit rate speech coding technique coupled with a relatively high degree of error protection may be employed. Systems can rapidly change between these different coding modes based upon varying changes in channel conditions.
As mentioned above, a mode indicator may be transmitted to the receiver (whether it be the base or mobile station's receiver) so that it can employ the appropriate channel decoding/speech decoding techniques. Typically, this mode indicator may include just a few, e.g., two, bits which are conveyed along with the data fields. Thus, it will be appreciated that it is particularly important for the receiver to be able to accurately decode the coding mode indicator since, otherwise, an entire frame of data may be unrecoverable. This desire for accurate reception of the mode indicator may lead designers to strongly protect the mode indicator with heavy channel coding.
However, usage of heavy channel coding implies higher redundancy, which means more bits to be transmitted for the mode indicator field. This is, as explained earlier, undesirable since overhead bits should be minimized, not increased. Thus, it would be desirable to provide techniques and systems for increasing the likelihood that mode indicators, such as the coding mode indicator, will be properly decoded, while at the same time minimizing the number of overhead bits which are transmitted with the payload data.